Adding packetsDiscarded to RTCReceivedRtpStreamStats.
Bug: webrtc:12532, webrtc:7065, webrtc:8199
Change-Id: I3ba62ec65e5660e98787f629aec3ee7a0889207a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/225261
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34468}
diff --git a/api/neteq/neteq.h b/api/neteq/neteq.h
index ea7079e..81340f1 100644
--- a/api/neteq/neteq.h
+++ b/api/neteq/neteq.h
@@ -60,7 +60,7 @@
// These metrics are never reset.
struct NetEqLifetimeStatistics {
// Stats below correspond to similarly-named fields in the WebRTC stats spec.
- // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
+ // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats
uint64_t total_samples_received = 0;
uint64_t concealed_samples = 0;
uint64_t concealment_events = 0;
diff --git a/api/stats/rtcstats_objects.h b/api/stats/rtcstats_objects.h
index 2030380..6995db8 100644
--- a/api/stats/rtcstats_objects.h
+++ b/api/stats/rtcstats_objects.h
@@ -399,13 +399,14 @@
// TODO(hbos) The following fields need to be added and migrated
// both from RTCInboundRtpStreamStats and RTCRemoteInboundRtpStreamStats:
- // packetsReceived, packetsDiscarded, packetsRepaired, burstPacketsLost,
+ // packetsReceived, packetsRepaired, burstPacketsLost,
// burstPacketDiscarded, burstLossCount, burstDiscardCount, burstLossRate,
// burstDiscardRate, gapLossRate, gapDiscardRate, framesDropped,
// partialFramesLost, fullFramesLost
// crbug.com/webrtc/12532
RTCStatsMember<double> jitter;
RTCStatsMember<int32_t> packets_lost; // Signed per RFC 3550
+ RTCStatsMember<uint64_t> packets_discarded;
protected:
RTCReceivedRtpStreamStats(const std::string&& id, int64_t timestamp_us);
@@ -463,8 +464,6 @@
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<double> round_trip_time;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
- RTCStatsMember<uint32_t> packets_discarded;
- // TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> packets_repaired;
// TODO(hbos): Collect and populate this value. https://bugs.webrtc.org/7065
RTCStatsMember<uint32_t> burst_packets_lost;
@@ -569,7 +568,7 @@
~RTCRemoteInboundRtpStreamStats() override;
// TODO(hbos): The following RTCReceivedRtpStreamStats metrics should also be
- // implemented: packetsReceived, packetsDiscarded, packetsRepaired,
+ // implemented: packetsReceived, packetsRepaired,
// burstPacketsLost, burstPacketsDiscarded, burstLossCount, burstDiscardCount,
// burstLossRate, burstDiscardRate, gapLossRate and gapDiscardRate.
// RTCRemoteInboundRtpStreamStats
diff --git a/audio/audio_receive_stream.cc b/audio/audio_receive_stream.cc
index f243fa6..f384381 100644
--- a/audio/audio_receive_stream.cc
+++ b/audio/audio_receive_stream.cc
@@ -298,6 +298,7 @@
// Get jitter buffer and total delay (alg + jitter + playout) stats.
auto ns = channel_receive_->GetNetworkStatistics(get_and_clear_legacy_stats);
+ stats.packets_discarded = ns.packetsDiscarded;
stats.fec_packets_received = ns.fecPacketsReceived;
stats.fec_packets_discarded = ns.fecPacketsDiscarded;
stats.jitter_buffer_ms = ns.currentBufferSize;
diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc
index fb5f1cb..1d6183a 100644
--- a/audio/audio_receive_stream_unittest.cc
+++ b/audio/audio_receive_stream_unittest.cc
@@ -68,8 +68,34 @@
123,
{"codec_name_recv", 96000, 0}};
const NetworkStatistics kNetworkStats = {
- 123, 456, false, 789012, 3456, 123, 456, 789, 543, 123, 432, 321, 123,
- 101, 789, 12, 345, 678, 901, 0, -1, -1, 0, 0, 0, 0};
+ /*currentBufferSize=*/123,
+ /*preferredBufferSize=*/456,
+ /*jitterPeaksFound=*/false,
+ /*totalSamplesReceived=*/789012,
+ /*concealedSamples=*/3456,
+ /*silentConcealedSamples=*/123,
+ /*concealmentEvents=*/456,
+ /*jitterBufferDelayMs=*/789,
+ /*jitterBufferEmittedCount=*/543,
+ /*jitterBufferTargetDelayMs=*/123,
+ /*insertedSamplesForDeceleration=*/432,
+ /*removedSamplesForAcceleration=*/321,
+ /*fecPacketsReceived=*/123,
+ /*fecPacketsDiscarded=*/101,
+ /*packetsDiscarded=*/989,
+ /*currentExpandRate=*/789,
+ /*currentSpeechExpandRate=*/12,
+ /*currentPreemptiveRate=*/345,
+ /*currentAccelerateRate =*/678,
+ /*currentSecondaryDecodedRate=*/901,
+ /*currentSecondaryDiscardedRate=*/0,
+ /*meanWaitingTimeMs=*/-1,
+ /*maxWaitingTimeMs=*/-1,
+ /*packetBufferFlushes=*/0,
+ /*delayedPacketOutageSamples=*/0,
+ /*relativePacketArrivalDelayMs=*/135,
+ /*interruptionCount=*/-1,
+ /*totalInterruptionDurationMs=*/-1};
const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
struct ConfigHelper {
@@ -253,6 +279,13 @@
EXPECT_EQ(static_cast<double>(kNetworkStats.jitterBufferTargetDelayMs) /
static_cast<double>(rtc::kNumMillisecsPerSec),
stats.jitter_buffer_target_delay_seconds);
+ EXPECT_EQ(kNetworkStats.insertedSamplesForDeceleration,
+ stats.inserted_samples_for_deceleration);
+ EXPECT_EQ(kNetworkStats.removedSamplesForAcceleration,
+ stats.removed_samples_for_acceleration);
+ EXPECT_EQ(kNetworkStats.fecPacketsReceived, stats.fec_packets_received);
+ EXPECT_EQ(kNetworkStats.fecPacketsDiscarded, stats.fec_packets_discarded);
+ EXPECT_EQ(kNetworkStats.packetsDiscarded, stats.packets_discarded);
EXPECT_EQ(Q14ToFloat(kNetworkStats.currentExpandRate), stats.expand_rate);
EXPECT_EQ(Q14ToFloat(kNetworkStats.currentSpeechExpandRate),
stats.speech_expand_rate);
@@ -264,6 +297,16 @@
stats.accelerate_rate);
EXPECT_EQ(Q14ToFloat(kNetworkStats.currentPreemptiveRate),
stats.preemptive_expand_rate);
+ EXPECT_EQ(kNetworkStats.packetBufferFlushes, stats.jitter_buffer_flushes);
+ EXPECT_EQ(kNetworkStats.delayedPacketOutageSamples,
+ stats.delayed_packet_outage_samples);
+ EXPECT_EQ(static_cast<double>(kNetworkStats.relativePacketArrivalDelayMs) /
+ static_cast<double>(rtc::kNumMillisecsPerSec),
+ stats.relative_packet_arrival_delay_seconds);
+ EXPECT_EQ(kNetworkStats.interruptionCount, stats.interruption_count);
+ EXPECT_EQ(kNetworkStats.totalInterruptionDurationMs,
+ stats.total_interruption_duration_ms);
+
EXPECT_EQ(kAudioDecodeStats.calls_to_silence_generator,
stats.decoding_calls_to_silence_generator);
EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
diff --git a/call/audio_receive_stream.h b/call/audio_receive_stream.h
index 8403e6b..182cd49 100644
--- a/call/audio_receive_stream.h
+++ b/call/audio_receive_stream.h
@@ -39,6 +39,7 @@
uint64_t fec_packets_received = 0;
uint64_t fec_packets_discarded = 0;
uint32_t packets_lost = 0;
+ uint64_t packets_discarded = 0;
uint32_t nacks_sent = 0;
std::string codec_name;
absl::optional<int> codec_payload_type;
diff --git a/media/base/media_channel.h b/media/base/media_channel.h
index 7b9a6f1..c6bbc07 100644
--- a/media/base/media_channel.h
+++ b/media/base/media_channel.h
@@ -471,7 +471,7 @@
int delay_estimate_ms = 0;
int audio_level = 0;
// Stats below correspond to similarly-named fields in the WebRTC stats spec.
- // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
+ // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats
double total_output_energy = 0.0;
uint64_t total_samples_received = 0;
double total_output_duration = 0.0;
@@ -483,6 +483,9 @@
uint64_t removed_samples_for_acceleration = 0;
uint64_t fec_packets_received = 0;
uint64_t fec_packets_discarded = 0;
+ // Stats below correspond to similarly-named fields in the WebRTC stats spec.
+ // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats
+ uint64_t packets_discarded = 0;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
// fraction of synthesized audio inserted through expansion.
float expand_rate = 0.0f;
diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
index aa80c87..7ce61b0 100644
--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
@@ -2365,6 +2365,7 @@
rinfo.fec_packets_received = stats.fec_packets_received;
rinfo.fec_packets_discarded = stats.fec_packets_discarded;
rinfo.packets_lost = stats.packets_lost;
+ rinfo.packets_discarded = stats.packets_discarded;
rinfo.codec_name = stats.codec_name;
rinfo.codec_payload_type = stats.codec_payload_type;
rinfo.jitter_ms = stats.jitter_ms;
diff --git a/modules/audio_coding/acm2/acm_receiver.cc b/modules/audio_coding/acm2/acm_receiver.cc
index 3214ce6..80cb3c5 100644
--- a/modules/audio_coding/acm2/acm_receiver.cc
+++ b/modules/audio_coding/acm2/acm_receiver.cc
@@ -307,6 +307,8 @@
neteq_->GetOperationsAndState();
acm_stat->packetBufferFlushes =
neteq_operations_and_state.packet_buffer_flushes;
+ acm_stat->packetsDiscarded =
+ neteq_operations_and_state.discarded_primary_packets;
}
int AcmReceiver::EnableNack(size_t max_nack_list_size) {
diff --git a/modules/audio_coding/include/audio_coding_module_typedefs.h b/modules/audio_coding/include/audio_coding_module_typedefs.h
index a7210da..e5598e3 100644
--- a/modules/audio_coding/include/audio_coding_module_typedefs.h
+++ b/modules/audio_coding/include/audio_coding_module_typedefs.h
@@ -81,19 +81,22 @@
// adding extra delay due to "peaky jitter"
bool jitterPeaksFound;
// Stats below correspond to similarly-named fields in the WebRTC stats spec.
- // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats
+ // https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats
uint64_t totalSamplesReceived;
uint64_t concealedSamples;
uint64_t silentConcealedSamples;
uint64_t concealmentEvents;
uint64_t jitterBufferDelayMs;
uint64_t jitterBufferEmittedCount;
- // Non standard stats propagated to spec complaint GetStats API.
- uint64_t jitterBufferTargetDelayMs;
uint64_t insertedSamplesForDeceleration;
uint64_t removedSamplesForAcceleration;
uint64_t fecPacketsReceived;
uint64_t fecPacketsDiscarded;
+ // Stats below correspond to similarly-named fields in the WebRTC stats spec.
+ // https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats
+ uint64_t packetsDiscarded;
+ // Non standard stats propagated to spec complaint GetStats API.
+ uint64_t jitterBufferTargetDelayMs;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
// fraction (of original stream) of synthesized audio inserted through
// expansion (in Q14)
diff --git a/pc/rtc_stats_collector.cc b/pc/rtc_stats_collector.cc
index 6599d0e..ee58264 100644
--- a/pc/rtc_stats_collector.cc
+++ b/pc/rtc_stats_collector.cc
@@ -392,6 +392,7 @@
voice_receiver_info.fec_packets_received;
inbound_audio->fec_packets_discarded =
voice_receiver_info.fec_packets_discarded;
+ inbound_audio->packets_discarded = voice_receiver_info.packets_discarded;
return inbound_audio;
}
diff --git a/pc/rtc_stats_collector_unittest.cc b/pc/rtc_stats_collector_unittest.cc
index 2ac0737..4ec86ab 100644
--- a/pc/rtc_stats_collector_unittest.cc
+++ b/pc/rtc_stats_collector_unittest.cc
@@ -1953,6 +1953,7 @@
cricket::SsrcReceiverInfo());
voice_media_info.receivers[0].local_stats[0].ssrc = 1;
voice_media_info.receivers[0].packets_lost = -1; // Signed per RFC3550
+ voice_media_info.receivers[0].packets_discarded = 7788;
voice_media_info.receivers[0].packets_rcvd = 2;
voice_media_info.receivers[0].nacks_sent = 5;
voice_media_info.receivers[0].fec_packets_discarded = 5566;
@@ -2009,6 +2010,7 @@
expected_audio.bytes_received = 3;
expected_audio.header_bytes_received = 4;
expected_audio.packets_lost = -1;
+ expected_audio.packets_discarded = 7788;
// |expected_audio.last_packet_received_timestamp| should be undefined.
expected_audio.jitter = 4.5;
expected_audio.jitter_buffer_delay = 1.0;
diff --git a/pc/rtc_stats_integrationtest.cc b/pc/rtc_stats_integrationtest.cc
index 2dfe1b5..df7b8a3 100644
--- a/pc/rtc_stats_integrationtest.cc
+++ b/pc/rtc_stats_integrationtest.cc
@@ -800,7 +800,9 @@
bool VerifyRTCInboundRTPStreamStats(
const RTCInboundRTPStreamStats& inbound_stream) {
RTCStatsVerifier verifier(report_, &inbound_stream);
- VerifyRTCReceivedRtpStreamStats(inbound_stream, verifier);
+ VerifyRTCReceivedRtpStreamStats(inbound_stream, verifier,
+ inbound_stream.media_type.is_defined() &&
+ *inbound_stream.media_type == "audio");
verifier.TestMemberIsOptionalIDReference(
inbound_stream.remote_id, RTCRemoteOutboundRtpStreamStats::kType);
if (inbound_stream.media_type.is_defined() &&
@@ -884,7 +886,6 @@
verifier.TestMemberIsUndefined(inbound_stream.frames_received);
}
verifier.TestMemberIsUndefined(inbound_stream.round_trip_time);
- verifier.TestMemberIsUndefined(inbound_stream.packets_discarded);
verifier.TestMemberIsUndefined(inbound_stream.packets_repaired);
verifier.TestMemberIsUndefined(inbound_stream.burst_packets_lost);
verifier.TestMemberIsUndefined(inbound_stream.burst_packets_discarded);
@@ -1022,16 +1023,23 @@
void VerifyRTCReceivedRtpStreamStats(
const RTCReceivedRtpStreamStats& received_rtp,
- RTCStatsVerifier& verifier) {
+ RTCStatsVerifier& verifier,
+ bool packets_discarded_defined) {
VerifyRTCRTPStreamStats(received_rtp, verifier);
verifier.TestMemberIsNonNegative<double>(received_rtp.jitter);
verifier.TestMemberIsDefined(received_rtp.packets_lost);
+ if (packets_discarded_defined) {
+ verifier.TestMemberIsNonNegative<uint64_t>(
+ received_rtp.packets_discarded);
+ } else {
+ verifier.TestMemberIsUndefined(received_rtp.packets_discarded);
+ }
}
bool VerifyRTCRemoteInboundRtpStreamStats(
const RTCRemoteInboundRtpStreamStats& remote_inbound_stream) {
RTCStatsVerifier verifier(report_, &remote_inbound_stream);
- VerifyRTCReceivedRtpStreamStats(remote_inbound_stream, verifier);
+ VerifyRTCReceivedRtpStreamStats(remote_inbound_stream, verifier, false);
verifier.TestMemberIsDefined(remote_inbound_stream.fraction_lost);
verifier.TestMemberIsIDReference(remote_inbound_stream.local_id,
RTCOutboundRTPStreamStats::kType);
diff --git a/stats/rtcstats_objects.cc b/stats/rtcstats_objects.cc
index a2d7aa0..6af724e 100644
--- a/stats/rtcstats_objects.cc
+++ b/stats/rtcstats_objects.cc
@@ -582,7 +582,8 @@
WEBRTC_RTCSTATS_IMPL(
RTCReceivedRtpStreamStats, RTCRTPStreamStats, "received-rtp",
&jitter,
- &packets_lost)
+ &packets_lost,
+ &packets_discarded)
// clang-format on
RTCReceivedRtpStreamStats::RTCReceivedRtpStreamStats(const std::string&& id,
@@ -593,13 +594,15 @@
int64_t timestamp_us)
: RTCRTPStreamStats(std::move(id), timestamp_us),
jitter("jitter"),
- packets_lost("packetsLost") {}
+ packets_lost("packetsLost"),
+ packets_discarded("packetsDiscarded") {}
RTCReceivedRtpStreamStats::RTCReceivedRtpStreamStats(
const RTCReceivedRtpStreamStats& other)
: RTCRTPStreamStats(other),
jitter(other.jitter),
- packets_lost(other.packets_lost) {}
+ packets_lost(other.packets_lost),
+ packets_discarded(other.packets_discarded) {}
RTCReceivedRtpStreamStats::~RTCReceivedRtpStreamStats() {}
@@ -650,7 +653,6 @@
&total_samples_duration,
&frames_received,
&round_trip_time,
- &packets_discarded,
&packets_repaired,
&burst_packets_lost,
&burst_packets_discarded,
@@ -706,7 +708,6 @@
total_samples_duration("totalSamplesDuration"),
frames_received("framesReceived"),
round_trip_time("roundTripTime"),
- packets_discarded("packetsDiscarded"),
packets_repaired("packetsRepaired"),
burst_packets_lost("burstPacketsLost"),
burst_packets_discarded("burstPacketsDiscarded"),
@@ -758,7 +759,6 @@
total_samples_duration(other.total_samples_duration),
frames_received(other.frames_received),
round_trip_time(other.round_trip_time),
- packets_discarded(other.packets_discarded),
packets_repaired(other.packets_repaired),
burst_packets_lost(other.burst_packets_lost),
burst_packets_discarded(other.burst_packets_discarded),
